[..] |
ASTERISK-28000: sample: PJSIP endpoint identifier order doesn't match reality |
ASTERISK-28001: res_pjsip_registrar: Improve performance of inbound handling |
ASTERISK-28002: When T.140 realtime text is negociated, a lot of debug traces are generated |
ASTERISK-28003: Qualifying non-authenticated endpoints on startup |
ASTERISK-28004: dns: Core ast_dns_get_nameservers does not support configured IPv6 servers |
ASTERISK-28005: channel.c: ARI ring only once |
ASTERISK-28006: PJSIP: Missing "party=calling"/"party=called" in Remote-Party-ID |
ASTERISK-28007: rtcp-mux is put in SDP answer regardless of offer |
ASTERISK-28008: Asterisk crash signal 11, Segmentation fault |
ASTERISK-28009: Queue predial for callee channel |
ASTERISK-28010: PJSIP: Crash with MWI implicit subscription replaced by explicit |
ASTERISK-28011: chan_sip: get_refer_info() attempted unlock mutex 'peer' without owning it! |
ASTERISK-28012: ODBC Voicemail and 'pollmailboxes=yes' does not update shared state via XMPP |
ASTERISK-28013: res_http_websocket: Crash when reading HTTP Upgrade requests |
ASTERISK-28014: Can't record video with Record application |
ASTERISK-28015: pjsip: bad file descriptor when passing pjsip qualify endpoint to standard out |
ASTERISK-28016: PJSIP sends duplicate 183 Progress responses |
ASTERISK-28017: I'm using Chan-Sip but PJSIP errors/warnings. |
ASTERISK-28018: IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate |
ASTERISK-28019: Crash in ast_format_get_sample_rate when play an audio |
ASTERISK-28020: res_pjsip_transport_websocket: Properly set 'received' for IPv6 |
ASTERISK-28021: backtrace.c: New crash due to double-free. |
ASTERISK-28022: res_pjsip realtime: uri column in ps_contacts table can be too short |
ASTERISK-28023: CONFBRIDGE maybe break when playing annoucement |
ASTERISK-28024: my queue members (eg. SIP/{SIPuser name}/{phone no}) showing invalid. Is it neccessary to registered all the agents of queue in sip.conf. Or please give me alternative .Please help me with the issue |
ASTERISK-28025: Asterisk webrtc in SIPML5 |
ASTERISK-28026: Session timers not updating after reload for active calls |
ASTERISK-28027: Call Setup Crash ILBC -> ULAW |
ASTERISK-28028: Disk I/O error, dropped calls |
ASTERISK-28029: [patch] res_musiconhold : music on hold will not start if previous hold just reached end of file |
ASTERISK-28030: pbx_lua: Deadlock when reloading module |
ASTERISK-28031: Invalid UTF-8 string - Element Block |
ASTERISK-28032: Realtime queuemembers are not updated during retry phase |
ASTERISK-28033: AMI event "NewExten" is set to the wrong class |
ASTERISK-28034: chan_sip unstable with TLS after asterisk start or reloads |
ASTERISK-28035: PJSIP Error |
ASTERISK-28036: Codec negotiation when incoming re-INVITE has no SDP |
ASTERISK-28037: How to set runing REST API after hungup a call |
ASTERISK-28038: Queue log is incorrect in Attended transfer &Blind transfer |
ASTERISK-28039: Null pointer crash for ast_stream_get_type |
ASTERISK-28040: pbx: "dialplan reload" is removing minus symbol from dynamic hints |
ASTERISK-28041: Asterisk freezes unexpected |
ASTERISK-28042: Asterisk freezes unexpected |
ASTERISK-28043: Error 4 in app_queue.so |
ASTERISK-28044: res_stasis : random crash related to ast_channel_varset_type |
ASTERISK-28045: configure script does not enforce libunbound2 version |
ASTERISK-28046: Remove stale nonoptreq references |
ASTERISK-28047: chan_pjsip: Declined video stream is added when no video codecs configured and session refresh with removed video stream occurs |
ASTERISK-28048: res_pjsip fails to migrate endpoint devstate from Unavailable to Not in use after restart until pjsip reload (or reregister) |
ASTERISK-28049: res_pjproject build failure |
ASTERISK-28050: Asterisk crash with Abort error |
ASTERISK-28051: RTP engine should only accept audio frames with allowed payloads |
ASTERISK-28052: app_voicemail: Voicemail help plays conflicting options |
ASTERISK-28053: chan_pjsip: Wrong or missing Q.850 reason in CANCEL |
ASTERISK-28054: Asterisk Core dumping on regular basis 13.21.1 |
ASTERISK-28055: app_queue: Per-member wrapup time missing from AddQueueMember application |
ASTERISK-28056: res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR |
ASTERISK-28057: chan_sip: SipNotify via AMI behaves differently to CLI |
ASTERISK-28058: Set busy on soft phone |
ASTERISK-28059: PJSIP: Update bundled PJPROJECT to version 2.8 |
ASTERISK-28060: Queue answered out of order |
ASTERISK-28061: Creating new realtime pjsip endpoint not updating state |
ASTERISK-28063: unable to register IP extension in FreePBX |
ASTERISK-28064: Memory leak issue in json.c, endpoints.c and stasis_channels.c While using AMI |
ASTERISK-28065: res_odbc: missing SQL error diagnostic |
ASTERISK-28066: modification of modules.conf |
ASTERISK-28067: res_pjsip_sdp_rtp: Extra fingerprint attribute in SDP |
ASTERISK-28068: Wrong button label in german dpma localization |
ASTERISK-28069: Dropping CDRs records with local languages |
ASTERISK-28070: testsuite: Sniffer assumes pjmedia will use ports below 10000 |
ASTERISK-28071: chan_sip: ignores the "fromdomain" option if "fromuser" option is presented |
ASTERISK-28072: app_agent_pool: Crash when heavily manipulated externally using AMI |
ASTERISK-28073: asterisk memory leak |
ASTERISK-28074: Run Two Exec command in asterisk application |
ASTERISK-28075: Null pointer in SRTP Handling Crash |
ASTERISK-28076: bridging: Asterisk crashes when receiving an empty realtime text frame |
ASTERISK-28077: res_pjsip: improve realtime performance on CLI 'pjsip show contacts' |
ASTERISK-28078: pjsip: Missing support for TLS CRL |
ASTERISK-28079: [patch]core: New variables CONNECTED_LINE_ORIGINAL_* for interception routine CONNECTED_LINE_SEND_SUB |
ASTERISK-28080: AsteriskNOW |
ASTERISK-28081: chan_sip: Asterisk 12+ chan_sip doesn't report AST_CEL_PICKUP in handle_invite_replaces |
ASTERISK-28082: res_ari_bridges: allow the app to be specified on bridge creation when using ARI |
ASTERISK-28083: res_agi: Asterisk truncates result of get_variable to 1024 characters |
ASTERISK-28084: app_queue: QueueMemberStatus Event flooding AMI |
ASTERISK-28085: testsuite: Figure out why chan_sip blind transfer tests are failing. |
ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI |
ASTERISK-28087: add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip |
ASTERISK-28088: ast_restart: Test is failing occasionally |
ASTERISK-28089: function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload |
ASTERISK-28090: chan_sip: MessageSend() doesn't use configured SIP peers |
ASTERISK-28091: Control endpoints media_address use per call basis |
ASTERISK-28092: res_pjsip. AMI event Registry. Field Cause is empty |
ASTERISK-28093: pbx: Deadlock from holding channel lock when it shouldn't be |
ASTERISK-28094: pjsip. Disable anonymous for local sip domains and force to inbound registration |
ASTERISK-28095: func_odbc: Crash when calling an ODBC function from another ODBC function |
ASTERISK-28096: pjsip: PJPROJECT 2.8 causes test failure |
ASTERISK-28097: Queue option 'b' + SIP_HEADER make issue |
ASTERISK-28098: ATTENDED_TRANSFER_COMPLETE_SOUND deadlocks in Local channels (Asterisk 11) |
ASTERISK-28099: When I try to find the state of Endpoints it is showing unavailable |
ASTERISK-28100: how to disable native_dahdi techonology for bridge |
ASTERISK-28101: Unable to load config file 'statsd.conf' |
ASTERISK-28102: stasis: Use implementation specific cache for channel snapshots |
ASTERISK-28103: stasis: Filter messages at publishing to reduce work done |
ASTERISK-28104: AstriCon Feedback: Automatically create a 1 line dialplan context for stasis apps |
ASTERISK-28105: AstriCon Feedback: Allow multiple connections for the same Stasis application |
ASTERISK-28106: Astricon Feedback: Unable to filter ARI events when GETting causes overload of events |
ASTERISK-28107: app_confbridge: Participant info labels aren't being added to the SDPs |
ASTERISK-28108: Deadlock in publish_msg (stasis.c) |
ASTERISK-28109: pbx_dundi: Does not support chan_pjsip |
ASTERISK-28110: rtp: Incorrect Packetization |
ASTERISK-28111: build: CHANGES/UPGRADE are irritating to work with. |
ASTERISK-28112: Asterisk is not able to use newly released mysql connector odbc 8.0.12 for voicemail |
ASTERISK-28113: gerrit: Minor Tweak to email template to support dark theme. |
ASTERISK-28114: Random crash |
ASTERISK-28115: rtp: Cache channel snapshot locally |
ASTERISK-28116: stasis: Investigate automatic disabling of message creation |
ASTERISK-28117: stasis: Add statistics for usage when in developer mode |
ASTERISK-28118: Not forwarding RTP packets / Packet loss |
ASTERISK-28119: stasis: Segment channel snapshot to reduce creation cost |
ASTERISK-28120: stasis: Audit loitering topics |
ASTERISK-28121: Don't play the early media when I have an incoming call |
ASTERISK-28122: app_queue.so crashed |
ASTERISK-28123: stun keeps revaluating |
ASTERISK-28124: Is Asterisk SAP Gateway supported in Pakistan, Are you offering a Cloud solution for SAP Gateway |
ASTERISK-28125: app_queue: Revert broken queue channel reference patch |
ASTERISK-28126: Can't login into asterisk.org |
ASTERISK-28127: Buffer overflow for DNS SRV/NAPTR records |
ASTERISK-28128: postgresql config db upgrade fail at alembic upgrade fe6592859b85 |
ASTERISK-28129: Incorrect Behavior for rewrite_contact when Re-Invite omits routset |
ASTERISK-28130: when calling from the queue, all contacts are not called |
ASTERISK-28131: ${ANSWEREDTIME} is incorrect in Dial app in Asterisk 15.6 |
ASTERISK-28132: res_pjsip_registrar: Asterisk crashing with large number of PJSIP registration |
ASTERISK-28133: Realtime iaxfriends table 'port' definition causing issues |
ASTERISK-28134: Legacy forums using Symantec certs throw warning on Chrome |
ASTERISK-28135: Opus Codec Parked Calls Drop |
ASTERISK-28136: Allow the sip_to_pjsip script to be used in a pipe |
ASTERISK-28137: res_pjsip_notify: improve realtime performance on CLI completion on the endpoint |
ASTERISK-28138: sip_to_pjsip.py does not convert setvar to set_var |
ASTERISK-28139: RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls |
ASTERISK-28140: repeated segmentation faults |
ASTERISK-28141: mysql same value calldate,answer,end |
ASTERISK-28142: res_agi: asterisk will execute continuation of agi file in situation of simultaneous hangup in both side (caller and callee) |
ASTERISK-28143: app_amd: Infinite loop on silent calls |
ASTERISK-28144: [patch] New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI |
ASTERISK-28145: SRTp with bria and zoiper |
ASTERISK-28146: pbx_config: Only the first [globals] section is processed. |
ASTERISK-28147: Unable to connect to Asterisk from asterisk-java |
ASTERISK-28148: MoH restart at each Dial |
ASTERISK-28149: PJSIP: Setting CallerID for outbound channel from predial handler doesn't work |
ASTERISK-28150: Formatting error in documentation |
ASTERISK-28151: app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default |
ASTERISK-28152: mysql two record |
ASTERISK-28153: [patch] chan_sip: fix Reason-Phrase for 603 Response |
ASTERISK-28154: stasis: Add support for shutting down topic |
ASTERISK-28155: res_pjsip_mwi: Crash at shutdown due to order problem |
ASTERISK-28156: Race condition involving session->media (res_pjsip_session) leads to crash. |
ASTERISK-28157: Asterisk crashes when the res_pjsip_* modules unload |
ASTERISK-28158: Some conditions prevent running of el_end, break the terminal. |
ASTERISK-28159: SIGABRT caused by stack corruption in hashkeys_read when no matching keys present |
ASTERISK-28160: Asterisk was failed down and restarted again. |
ASTERISK-28161: Removal of Previous Patch Causes PJSIP Timer Issues |
ASTERISK-28162: [patch] need to reset DTMF last sequence number and timestamp on RTP renegotiation |
ASTERISK-28163: Asterisk with 2 nic's |
ASTERISK-28164: stasis: Improve channel snapshot segmenting |
ASTERISK-28165: app_queue: QueueMemberStatus ami event duplication |
ASTERISK-28166: app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC |
ASTERISK-28167: 256 cipher during outgoing calls |
ASTERISK-28168: app_queue: Adding a blank entry into sql queue_members crashes asterisk. |
ASTERISK-28169: ARI /channels/create handler causes core dump |
ASTERISK-28170: ARI POST /channels/{channelId}/dial memory leak |
ASTERISK-28171: res_resolver_unbound: DNS issue when under load, can't make outgoing calls |
ASTERISK-28173: Deadlock in chan_sip handling subscribe request during res_parking reload |
ASTERISK-28174: PJSIP issue with TEL (RFC 3966) |
ASTERISK-28175: PJSIP support for TEL (RFC 3966) |
ASTERISK-28176: Make usage tracking switchable |
ASTERISK-28177: block calling from contextA to contextB |
ASTERISK-28178: Program terminated with signal 11, Segmentation fault. |
ASTERISK-28179: Asterisk responses 100 trying about 4 seconds after sending INVITE. |
ASTERISK-28180: ami: High memory (increased upto 15 GBs) while pushing 500 calls continuously for 7 hours |
ASTERISK-28181: ari: Originating overwrites channel start time |
ASTERISK-28182: chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE |
ASTERISK-28183: Channel SIP message notifier |
ASTERISK-28184: Asterisk hangs up receiving unexpected frame format |
ASTERISK-28185: chan_pjsip: Subsequent same responses are not stopped |
ASTERISK-28186: stasis: Filter messages at publishing based on to_* presence |
ASTERISK-28187: Asterisk 16.0 crash on receiving fax |
ASTERISK-28188: mysql db import error |
ASTERISK-28189: configured pjsip endpoints go offline when a new endpoint registers |
ASTERISK-28190: Asterisk Crashing |
ASTERISK-28191: features.conf option blindxfer doesn't seem to do Pattern Matching. |
ASTERISK-28192: Use kamailio and asterisk on call about 500 current |
ASTERISK-28193: auto gain control |
ASTERISK-28194: chan_sip: Leak using contact ACL |
ASTERISK-28195: stasis: Don't create subscription change messages if noone cares |
ASTERISK-28196: bridge_softmix: Does not support WebRTC source with multi video tracks. |
ASTERISK-28197: stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases |
ASTERISK-28198: res_ari: Add new hangup causes for ARI Channel DELETE command |
ASTERISK-28199: sdp: iLBC codec does not contain "mode=" attribute |
ASTERISK-28200: res_rtp_asterisk: Duplicate DTMF with endpoint when media received in between |
ASTERISK-28201: [patch] confbridge: no announce to the marked users when they join an empty conference |
ASTERISK-28202: pjproject: fails to build on ppc64el |
ASTERISK-28203: Asterisk 13.13 crash/restart bug |
ASTERISK-28204: Asterisk Restarted | Crashes chan_sip.c |
ASTERISK-28205: module app_queue.so stoped |
ASTERISK-28206: Read Application is NOT correct in single channel |
ASTERISK-28207: promiscredir |
ASTERISK-28208: chan_pjsip: 183 without SDP followed by 180 does not result in media |
ASTERISK-28209: res_pjsip_t38: Passthrough causes crash when re-invite collision |
ASTERISK-28210: res_pjsip_outbound_registration: Registrations reports "Rejected" for No Response type failures |
ASTERISK-28211: chan_pjsip: Path header is not used with PJSIP_DIAL_CONTACTS |
ASTERISK-28212: stasis: Statistics broke ABI under developer mode |
ASTERISK-28213: res_pjsip: Threads pile up needlessly when AOR is blocked |
ASTERISK-28214: PJSIP and CHAN_SIP issues |
ASTERISK-28215: app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs |
ASTERISK-28216: Crash when race condition between manager_play_dtmf and ast_hangup |
ASTERISK-28217: mwi broken? |
ASTERISK-28218: app_queue: Asterisk crashes when using Queue with a pre-dial handler (option b) |
ASTERISK-28219: res_ari: Channel create and dial may cause "BUG! Must supply a channel name.." error |
ASTERISK-28220: SPY extension follow the channel after being bridged |
ASTERISK-28221: Bug in ast_coredumper |
ASTERISK-28222: Regression: MWI polling no longer works |
ASTERISK-28223: Configure failure |
ASTERISK-28224: res_parking: ParkAndAnnounce hangs up call when lot is full. |
ASTERISK-28225: app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked "urgent" |
ASTERISK-28226: I want to know about how to connect mysql to store CDR |
ASTERISK-28227: Adding more ARI subscription type |
ASTERISK-28228: res_pjsip: pjsip show contacts prints double entries |
ASTERISK-28229: Asterisk not responding/reloading |
ASTERISK-28230: res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony |
ASTERISK-28231: res_http_websocket: Not responding to Connection Close Frame (opcode 8) |
ASTERISK-28232: core: RAII using clang use-after-scope issue |
ASTERISK-28233: pbx_dundi: PJSIP is not a supported technology |
ASTERISK-28234: pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi |
ASTERISK-28235: ERROR[-1]: app_voicemail.c:2836 inboxcount2: Couldn't find mailbox in context |
ASTERISK-28236: Support separated HTTP request |
ASTERISK-28237: "FRACK!, Failed assertion bad magic number" happens when unsubscribe an application from an event source |
ASTERISK-28238: PJSIP realtime. getcontext not working with DUNDI |
ASTERISK-28239: Device_state - Change of returned status |
ASTERISK-28240: Unexpected unhold when Asterisk cannot find the moh-files |
ASTERISK-28241: Call pickup fails, if dialed from subroutine, but succeeds with macro |
ASTERISK-28242: Can't Retrieve Voicemail from PostgreSQL database when "msgnum" is INT type |
ASTERISK-28243: Corrupted SIP after handling a 302 redirect |
ASTERISK-28244: stasis: Filter messages at publishing to AMI/ARI |
ASTERISK-28245: DMTF emulation not working if direct_media = yes and dtmf_mode = info is set |
ASTERISK-28246: Support skipping on the g726 format |
ASTERISK-28247: res_ari: Applications not being cleaned up after certain scenarios |
ASTERISK-28248: testsuite: Write tests for ARI 'move' |
ASTERISK-28249: res_monitor: Segfault with Monitor(wav,file,i) |
ASTERISK-28250: build: Cross-compilation fails for target arm-linux-gnueabihf |
ASTERISK-28251: CI: Fix CI so it reverifies commit message changes |
ASTERISK-28252: HangupHandler manager events are never thrown |
ASTERISK-28253: res_pjsip_session: Adding rtcp stats result into the session |
ASTERISK-28254: testsuite: PJSIP tests can't tolerate retransmissions (and they happen sometimes) |
ASTERISK-28255: res_rtp_asterisk: REMB RTCP packet sending may be incorrect |
ASTERISK-28256: Video plays back in slow motion |
ASTERISK-28257: res_http_websocket: PING / PONG opcodes break data reception |
ASTERISK-28258: DUNDi Does Not Register chan_pjsip Realtime Endpoints On Register |
ASTERISK-28259: CLONE - app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs |
ASTERISK-28260: Asterisk segfault when rtp negotiation is wrong or fails |
ASTERISK-28261: PJSIP: SDP in 180 Ringing is ignored |
ASTERISK-28262: Custom device state losing - (dash) |
ASTERISK-28263: codec_opus: errors setting max_playback_rate and bitrate to "sdp" |
ASTERISK-28264: Added topic_all container |
ASTERISK-28265: PJSIP show channelstats incorrect information output |
ASTERISK-28266: Added ARI resource /ari/aserisk/ping |
ASTERISK-28267: res_stasis: Add ability to switch applications |
ASTERISK-28268: Asterisk + create new app + conncet mysql |
ASTERISK-28269: chan_mobile can't connect to phone again |
ASTERISK-28270: Creating call using ARI, does not increasing/decreasing the call count(core show calls) |
ASTERISK-28271: Opensuse Leap 15 --with-jannson-bundled will not compile |
ASTERISK-28272: The basic-pbx config samples don't produce a running asterisk |
ASTERISK-28273: H245 logical channels don't close when asterisk is terminated the call. |
ASTERISK-28274: Asterisk 11 crashes randomly |
ASTERISK-28275: Error saving agent on queue_log on pickup calls with * 8 |
ASTERISK-28276: TESTTIME feature not working |
ASTERISK-28277: database: Add some basic logging |
ASTERISK-28278: Asterisk extensions periodically lost registrations |
ASTERISK-28279: Added creation timestamp for bridge |
ASTERISK-28280: chan_sip problem with registration when challenge contains a "domain" field with protocol. |
ASTERISK-28281: Enabling cli command bridge destroy <bridge id> |
ASTERISK-28282: AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) |
ASTERISK-28283: Fixed wrong description for RTCPReceived/RTCPSent |
ASTERISK-28284: switching between native_bridge and simple_bridge can cause one way audio |
ASTERISK-28285: Fixed wrong RTT calculation |
ASTERISK-28286: chan_sip - no lock pvt data in proc_session_timer() |
ASTERISK-28287: Pjsip rewrite port in from/to headers on reply |
ASTERISK-28288: Resources (udptl fd) leaking for T.38 calls |
ASTERISK-28289: Feature: Allow detection of inband progress for outbound channels |
ASTERISK-28290: res_resolver_unbound.so: Failed to perform async DNS resolution |
ASTERISK-28291: res_pjsip_path: Not applied on OPTIONS requests |
ASTERISK-28292: Changed to show all channel stats including wrong media |
ASTERISK-28293: New Build ast_expr2fz.o Will Not Link |
ASTERISK-28294: Segmentation Fault on strchr |
ASTERISK-28295: chan_sip / pjsip: Non UTF-8 handling could be better |
ASTERISK-28296: Getting Error While Running asterisk -rvvv |
ASTERISK-28297: cdr_engine - taskprocessor.c |
ASTERISK-28298: chan_iax2: Does not update endpoint state in all cases |
ASTERISK-28299: 481 Call/Transaction Does Not Exist on receiving MESSAGE event |
ASTERISK-28300: AST_PBX_MAX_STACK is too low for some applications |
ASTERISK-28301: Allow voicemail boxes to be subscribed to with a presence event package |
ASTERISK-28302: ARI: "Error destroying mutex" when listing all ARI applications |
ASTERISK-28303: res_rtp_asterisk: Interaction between smoother and DTMF can cause out of order timestamps |
ASTERISK-28304: app_voicemail. Issue with NOTIFYs |
ASTERISK-28305: register_aor_core: Unable to bind contact |
ASTERISK-28306: res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent |
ASTERISK-28307: Segmentation fault in libasteriskpj.so.2 |
ASTERISK-28308: Double line at pjsip show contacts |
ASTERISK-28309: res_pjsip: Wrong Contact and Via fields with multiple UDP interfaces |
ASTERISK-28310: Inconsistent Stasis/AMI event ordering for queue members calling |
ASTERISK-28311: dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format |
ASTERISK-28312: res_pjsip_diversion: Corrupted SIP Diversion field after handling a 302 redirect |
ASTERISK-28313: ARI: userevents should be delivered via AMI too |
ASTERISK-28314: ARI: API changed but "apiVersion" in rest-api\resources.json did not |
ASTERISK-28315: Asterisk is crash in Amazon EC2, i recompile with --disable BUILD_NATIVE, but is crash anyway |
ASTERISK-28316: Asterisk Restarting / Creating core dump file |
ASTERISK-28317: Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function |
ASTERISK-28318: Conference Bridge Options (via AMI) Setting Bug |
ASTERISK-28319: musl: Crash on startup when loading modules |
ASTERISK-28320: Added ARI resource /ari/channels/{channelid}/rtp_statistics |
ASTERISK-28321: res_rtp_asterisk: Fixing possible divide by zero for rtcp stat calculation |
ASTERISK-28322: chan_pjsip: Add option to allow ignoring of 183 without SDP |
ASTERISK-28323: pjsip: sip.conf to pjsip.conf conversion script fails |
ASTERISK-28324: res_mwi_devstate does not allow variable replacement in dialplan |
ASTERISK-28325: Command to delete a bridge |
ASTERISK-28326: ari: Added timestamp for some ari events. |
ASTERISK-28327: res_pjsip_endpoint_identifier_ip: Missing IdentifyDetail event |
ASTERISK-28328: MeetMe global non-admin mute is muting admins that subsequently join |
ASTERISK-28329: RTCP - Error building JSON |
ASTERISK-28330: call gets disconnected every 30 sec after answer |
ASTERISK-28331: Add an option in MeetMe to start the conference with all non-admins muted |
ASTERISK-28332: Variable ALTCONF ignored when service is used in Debian |
ASTERISK-28333: StasisEnd event makes wrong timestamp value |
ASTERISK-28334: pjsip : direct_media options doesnt work |
ASTERISK-28335: stasis: Make topic and maybe subscription names unique and more useful |
ASTERISK-28336: After ARI continue, hangup() application does not create SoftHangupRequest event |
ASTERISK-28337: Regular segmentation faults |
ASTERISK-28338: Asterisk crashes with ERROR *** /usr/sbin/asterisk': corrupted size vs. prev_size: 0x00007f77400cfcf0 *** when there are 48 SIP outbound calls.(almost 144 SIP Channels) |
ASTERISK-28339: Added ARI resource /ari/channels/{channelid}/dump |
ASTERISK-28340: MWI NOTIFY is not sent immediately after a VM was created/deleted |
ASTERISK-28341: res_config_odbc eliminates empty custom (“@” prefix) variables |
ASTERISK-28342: Ast-to-Ast setup using the same rtcpinterval crashes RTCP and audio stream |
ASTERISK-28343: Added app_name, app_data to channel type |
ASTERISK-28344: Wrong music on hold handling on multi party attendant transfer |
ASTERISK-28345: IMS TEL URI incoming INVITE RFC 3966 not recognized |
ASTERISK-28346: useless transcoding |
ASTERISK-28347: ari: Crash while deleting bridge with channels in it |
ASTERISK-28348: Failed to initialize OOH323 endpoint-OOH323 Disabled |
ASTERISK-28349: Pause reason not reported in QueueMember AMI event |
ASTERISK-28350: manager: Stasis backed up due to locking |
ASTERISK-28351: CDR disposition incorrectly logged at file when AMD classifies as MACHINE |
ASTERISK-28352: German language sounds referenced by app-voicemail missing |
ASTERISK-28353: stasis: Crash at shutdown when statistics enabled |
ASTERISK-28354: app_queue: Call to Unavailable member when ringinuse=yes and another member is available |
ASTERISK-28355: Trunk to Inbound works for a few seconds only |
ASTERISK-28356: app_queue: CLI set ringinuse for realtime member not working |
ASTERISK-28357: Fixing duplicated subscription adding |
ASTERISK-28358: Too much allocations in frame.c |
ASTERISK-28359: User login page does not apply any form of price determination |
ASTERISK-28360: User login page does not apply any form of price determination |
ASTERISK-28361: app_confbridge: Missing MIXMONITOR_FILENAME in ConfBridge AMI |
ASTERISK-28362: strtok_r() makes gcc compile warning |
ASTERISK-28363: Millisecond-resolution call stats including PDD in channel variables |
ASTERISK-28364: func_strings: HASHKEYS in shared variable space cannot be retrieved |
ASTERISK-28365: New ARI for application execution. |
ASTERISK-28366: Add timeout for response to StasisStart-event |
ASTERISK-28367: Servname not supported for ai_socktype / Could not resolve socket address |
ASTERISK-28368: Low performance. Many errors taskprocessor |
ASTERISK-28369: app_queue: Member device state "invalid" when second call is ringing and hint is used |
ASTERISK-28370: res_pjsip_t38: Not accepting Audio Re-invite after T.38 rejection |
ASTERISK-28371: chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info |
ASTERISK-28372: Asterisk REPLY Wrong Contact header port (TCP) |
ASTERISK-28373: persistant registrations not working for TCP transport |
ASTERISK-28374: latest asterisk unconditionally launch gcc --version, even if the compiler is different |
ASTERISK-28375: res_pjsip: New configuration setting to allow disabling norefersub |
ASTERISK-28376: After changing parameter "media_address" in sip.conf to any valid IP, we are getting one way or no voice issues on few calls. Sometimes voice passes successfully, we would like to know why this is happening and resolution for same. |
ASTERISK-28377: ARI: Crash when unanswered channel rejoins dial bridge automatically |
ASTERISK-28378: Added detail subscriber/subscription info for stasis show app cli |
ASTERISK-28379: pjsip: show channelstats incorrect information output |
ASTERISK-28380: Asterisk CLI Showing ERRORS: frame.c: Excessive refcount 100000 reached on ao2 object when total calls reached almost 30000 count. |
ASTERISK-28381: MWI indicators not updating correctly on version 15.7.2 |
ASTERISK-28382: app_confbridge: Leave message not played when penultimate person leaves |
ASTERISK-28383: Contact status is reported as REACHABLE when contact is deleted |
ASTERISK-28384: res_xmpp: Crash when distribute_events=yes and res_mwi_devstate loads |
ASTERISK-28385: res_ari_channels: Added detail hangup code settings |
ASTERISK-28386: res_pjsip does not follow DNS SRV priority values |
ASTERISK-28387: res_pjsip: Contact status latency is not pushed through the AMI |
ASTERISK-28388: Endpoint sync causes device unreach when a new contact is added |
ASTERISK-28389: PJ ICE Rx error status code: 370400 'Bad Request'. |
ASTERISK-28390: Duplicate contacts appear when running "asterisk -rx 'pjsip list contacts'" |
ASTERISK-28391: res_indications: Crash requesting autocomplete on indications cli command |
ASTERISK-28392: The no-partial-inlining flag isn't passed to the bundled pjproject or jansson builds |
ASTERISK-28393: Multidomain support issue |
ASTERISK-28394: sip_outbound_publish_client_add_publisher Failed assertion bad magic number |
ASTERISK-28395: Asterisk occasionally fails to hangup channels |
ASTERISK-28396: missing voicemail option in sql table for realtime |
ASTERISK-28397: ARI Push Configuration - duplicate entries |
ASTERISK-28398: Column order of contrib realtime MySQL config for sippeers causes issues with NAT |
ASTERISK-28399: channel.c: Exceptionally long queue length queuing |
ASTERISK-28400: res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for transport-cc |
ASTERISK-28401: app_confbridge: Add *_all remb behavior variants |
ASTERISK-28402: res_pjsip_registrar: SEGV in registrar_find_contact |
ASTERISK-28403: Add native Prometheus support to Asterisk |
ASTERISK-28404: astobj2.c: FRACK, memory leakage |
ASTERISK-28405: ChanSpy : cannot hangup spying channel if no audio is received |
ASTERISK-28406: PJSIP compose invalid request-URI in ACK for re-INVITE |
ASTERISK-28407: Segfault in hash_ao2_find_next at astobj2_hash.c:581 |
ASTERISK-28408: Asterisk crashes intermittently if use sipML5 as the SIP client, regardless of the total number of peers in use. |
ASTERISK-28409: unexpected build problem at install procedure |
ASTERISK-28410: Crash on unload res_pjsip_mwi.so and show subscriptions |
ASTERISK-28411: Build link error bundled pjproject : relocation against symbol cant be used when shared object |
ASTERISK-28412: GCC 9 catches more string formatting issues |
ASTERISK-28413: pjsip show channelstats crashes asterisk while printing a channel being hung up |
ASTERISK-28414: Asterisk crash on internal calls |
ASTERISK-28415: segfault: sprint_list_entry (entry=entry@entry=0x7f9e30b4d8b0, line=line@entry=0x7f9e70676590 "\340ggp\236\177", len=256) at res_pjsip_history.c:669 |
ASTERISK-28416: Unable to get rtp codec payload code for slin |
ASTERISK-28417: SDP negotiation issue |
ASTERISK-28418: Timezone Problem |
ASTERISK-28419: app_amd: Does not work with silence suppression |
ASTERISK-28420: In WebRTC video call scenario, packet loss lead to frozen video。 |
ASTERISK-28421: Wrong type used for timestamp in res_rtp_asterisk |
ASTERISK-28422: Memory Leak in Confbridge menu |
ASTERISK-28423: ARI causes STASIS Deadlock |
ASTERISK-28424: Task processor queue regularly fills up with subscriptions, using curl config and external mwi, test phone doesn't stay registered |
ASTERISK-28425: Realtime Voicemail locks Asterisk when no filesystem folder exists |
ASTERISK-28426: Address out of bounds in ast_str_hash |
ASTERISK-28427: new mwi.h include missing from some dahdi source files, causes build failure |
ASTERISK-28428: app_dial: Incorrectly alters Hangup and DialEnd events when the c argument is passed, but app_dial didn't cancel the call. |
ASTERISK-28429: Bad answer JSON when request ARI |
ASTERISK-28430: res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF |
ASTERISK-28431: Asterisk memory increases and CentOS OEM killer kills Asterisk process |
ASTERISK-28432: ODBC |
ASTERISK-28433: More than 1 AMI connection ends up dying |
ASTERISK-28434: Segfault: INTERNAL_OBJ (user_data=0xffffffff) at astobj2.c:131 |
ASTERISK-28435: cdr_pgsql: Unix socket doesn't work |
ASTERISK-28436: Transcoding happening even though it is not necessary |
ASTERISK-28437: Taskprocessor doesn't process the tasks |
ASTERISK-28438: chan_pjsip: segfault in pj_grp_lock_acquire |
ASTERISK-28439: When we dial from a number the phone routes to a phone on the system instead of the number we dial |
ASTERISK-28440: pjsip: configure does not detect LibreSSL |
ASTERISK-28441: fax: T38 fallback to voice does not change codec |
ASTERISK-28442: stasis_state: Create a stasis module to cache last known state |
ASTERISK-28443: app_voicemail: remove dependency on stasis cache |
ASTERISK-28444: chan_pjsip: Peer IP for SSL handshake errors not logged |
ASTERISK-28445: res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled |
ASTERISK-28446: Asterisk NOTIFY NOT_INUSE and UNAVAILABLE both send same XML dialog <state>terminated</state> |
ASTERISK-28447: res_pjsip_messaging: In-dialog MESSAGE with no body causes crash |
ASTERISK-28448: chan_sip: Sometimes G729 (without annexb=no) is negotiated |
ASTERISK-28449: scheduler: Supported time can be exceeded by PUBLISH |
ASTERISK-28450: Program terminated with signal 11, Segmentation fault at t38_gateway.c:2189 |
ASTERISK-28451: chan_pjsip: HANGUPCAUSE(<channel>,tech) fails to get SIP cause for rejected calls |
ASTERISK-28452: pjsip: <sess-version> of SDP is not incremented though SDP may be changed on reinvite without SDP offer |
ASTERISK-28453: Voicemail: Failed to Lock Path: File Exists |
ASTERISK-28454: res_fax.c UTF-8 validation for remotestationid and pbx_builtin_setvar_helper |
ASTERISK-28455: res_odbc: Connection through proxysql fails |
ASTERISK-28456: Asterisk crashed and core dumps when attempting to free a frame |
ASTERISK-28457: [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-28317 |
ASTERISK-28458: res_pjsip_sdp_rtp: Remove unused variable |
ASTERISK-28459: pbx: Asterisk is dropping part of a string passed to functions |
ASTERISK-28460: res_pjsip_sdp_rtp: Fix ICE candidate leak with specific usage |
ASTERISK-28461: Crash in app.Pickup |
ASTERISK-28462: func_talkdetect: TALK_DETECT firing immediately even if phone microphone is muted |
ASTERISK-28463: res_pjsip_path: Crash when invalid contact is configured |
ASTERISK-28464: FRACK!, Failed assertion with res_pjsip |
ASTERISK-28465: Broken SDP can cause a segfault in a T.38 reINVITE |
ASTERISK-28466: Typo on the IVR System webpage |
ASTERISK-28467: 404 Link on Asterisk Wiki Dashboard |
ASTERISK-28468: 404 on What is Asterisk webpage |
ASTERISK-28469: AMI frozen on high load |
ASTERISK-28470: Mutex deadlock in audio_audiohook_write_list |
ASTERISK-28471: I have dial IVR number using asterisk AMI how can i send DTMF automatically |
ASTERISK-28472: Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV |
ASTERISK-28473: res_pjsip_t38: Crash on Asterisk 16.4 |
ASTERISK-28474: Old Book link on Asterisk Books webpage |
ASTERISK-28475: Segmentation fault |
ASTERISK-28476: Asterisk Deadlock During chan_pjsip Call Transfer |
ASTERISK-28477: Crash when not specifying "dbfile" in res_config_sqlite3.conf |
ASTERISK-28478: Crash performing "core reload" with modified res_config_sqlite3.conf |
ASTERISK-28479: Crash when no database specified using driver "sqlite3" in extconfig.conf |
ASTERISK-28480: json integer overflow in ssrc and timestamp |
ASTERISK-28481: Push notification using Asterisk server |
ASTERISK-28482: Asterisk 13.22.0 Segmentation fault PJSIP TLS+SRTP about 60 endpoints |
ASTERISK-28483: packet lost on UDPTL wrap around |
ASTERISK-28484: Add AudioSocket support |
ASTERISK-28485: Program terminated with signal 11, Segmentation fault. |
ASTERISK-28486: Randomly generated segfault in asterisk process |
ASTERISK-28487: compile menuselect on gentoo |
ASTERISK-28488: pjsip mwi: n+1 sip notify's sent on re-register |
ASTERISK-28489: Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain |
ASTERISK-28490: Segfault in chan_pjsip in grp_lock_dec_ref |
ASTERISK-28491: Allow in and out file descriptors to be used in AGI - Create XAGI |
ASTERISK-28492: pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group |
ASTERISK-28493: Can't log into community.asterisk.org |
ASTERISK-28494: Facing Segmentation fault (core dumped) identified by "asterisk -cvvvvvvv". I am also unable to connect remote asterisk because of this issue. Please provide solution. Thank you |
ASTERISK-28495: res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash |
ASTERISK-28496: PJSIP max_retries |
ASTERISK-28497: func_odbc: truncating Unicode string on readsql |
ASTERISK-28498: cel / cdr: Event times may be incorrect |
ASTERISK-28499: translate: Crash when frame does not have a "src" field set |
ASTERISK-28500: CDR Endtime is coming lesserthan the CDR Starttime |
ASTERISK-28501: Can't log into community.asterisk.org |
ASTERISK-28502: chan_pjsip incorrectly re-writes REGISTER 200 Response Contact |
ASTERISK-28503: Asterisk sudden crashes with segmentation fault |
ASTERISK-28504: Asterisk is crashing too frequently whenever a large number of PJSIP AOR are trying to register on asterisk. |
ASTERISK-28505: app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream |
ASTERISK-28506: asterisk crashes at random frequent intervals |
ASTERISK-28507: Wiki docs missing for MessageWaiting |
ASTERISK-28509: PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters |
ASTERISK-28510: Asterisk crashing on dtls handshake |
ASTERISK-28511: codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 |
ASTERISK-28512: Add pass-through support for H.265 (HEVC) codec |
ASTERISK-28513: Should To: be rewritten when forwarding to a phone |
ASTERISK-28514: phoneprov and RealTime |
ASTERISK-28515: res_pjsip: TLS close notify alert is not sent before closing the connection |
ASTERISK-28516: Crash Under AMI Taskprocessor Backup |
ASTERISK-28517: Asterisk segfault in t38_interpret_parameters at res_pjsip_t38.c:457 |
ASTERISK-28518: chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold |
ASTERISK-28519: [issue] Asterisk |
ASTERISK-28520: Failed to create temporary storage |
ASTERISK-28521: pjsip: Memory Leak |
ASTERISK-28522: chan_pjsip does not support fallback from t.38 to fax over alaw/ulaw |
ASTERISK-28523: Asterisk 16.5.0 Memory leak |
ASTERISK-28525: chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up |
ASTERISK-28526: Error executing SQL (COMMIT): database is locked |
ASTERISK-28527: ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf |
ASTERISK-28528: Local channel video stream broken with ConfBridge and first_marked=yes |
ASTERISK-28529: Segfault in res_pjsip_pubsub.c due to accessing a destroyed dialog |
ASTERISK-28530: Help in making Simultaneous calls NOT Bridged |
ASTERISK-28531: fetch all incoming calls mobile number to our extenstions |
ASTERISK-28532: Segfault at res_rtp_multicast.c:211 (function set_type) |
ASTERISK-28533: func_jitterbuffer: Add support for video synchronization |
ASTERISK-28534: Segmentation fault when there is no priority for an extension |
ASTERISK-28535: Error when change my callerid(num) |
ASTERISK-28536: Asterisk release candidates fail to build on FreeBSD |
ASTERISK-28537: Different music on hold queue, 1 for ringing and 1 for putting customer on hold |
ASTERISK-28538: chan_pjsip: Deadlock on fax detection |
ASTERISK-28539: Failed t.38 negotiation when B leg sends t.38 re-invite |
ASTERISK-28540: Deadlock In Stasis/ARI |
ASTERISK-28541: Asterisk 16.5.0 Memory leak |
ASTERISK-28542: [patch] add the ability for asterisk to generate on-hold re-invites |
ASTERISK-28543: When Asterisk cannot connect to SIP socket it starts to flood with "Bad descriptor" errors and hangs |
ASTERISK-28544: Wrong contact representation in ipv6 mode |
ASTERISK-28545: Introduce a 'playlist' mode for res_musiconhold |
ASTERISK-28546: MWI Leaks |
ASTERISK-28547: PJSIP AOR configuration "contact=sip:" crash |
ASTERISK-28548: PJSIP received a 183 with sendonly will be onhold |
ASTERISK-28549: Two repeated 183 |
ASTERISK-28550: SDP maxptime |
ASTERISK-28551: IPv4 address in SDP o= is (null) when configured for NAT using pjsip |
ASTERISK-28552: res_pjsip_mwi: Frack during unload on unsolicited_mwi container |
ASTERISK-28553: stasis.c: Crash during unload |
ASTERISK-28554: [patch] Add recipes for sample Queues |
ASTERISK-28555: Unable to Register WebRTC client when using a Proxy |
ASTERISK-28556: pjsip blind transfer fails |
ASTERISK-28557: dtmf detection |
ASTERISK-28558: BridgeAttendedTransfer not received if audio is playing in a bridge |
ASTERISK-28559: Making Asterisk work with Amtelco Genesis Software |
ASTERISK-28560: do_monitor Thread hangs on 99% cpu and doesn't respond |
ASTERISK-28561: Asterisk Deadlocks |
ASTERISK-28562: SIP WSS message not processed until next frame arrives |
ASTERISK-28563: Additional configuration [extName](+) not always working |
ASTERISK-28564: Memory leak with pjsip 2.9 and SIPS / SRTP |
ASTERISK-28565: Conference is disconnecting after entering conference PIN number |
ASTERISK-28566: CDR backend unload problem during active call(s) |
ASTERISK-28567: Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup. |
ASTERISK-28568: Potential memory leak on core reload |
ASTERISK-28569: Missing check for variable buf in function config_text_file_load in utils/extconf.c |
ASTERISK-28570: Potential infinite loop in function find_matching_priority |
ASTERISK-28571: cdr_pgsql: accesses obsolete (and finally removed) column |
ASTERISK-28572: Memory leaks in res_calendar_exchange and res_calendar_icalendar |
ASTERISK-28573: Missing event AgentComplete on AttendedTransfer |
ASTERISK-28574: pjproject fails to build on 16.6.0, works on 16.5 |
ASTERISK-28575: MWI Send Notify Crash on 16.6 |
ASTERISK-28576: res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match |
ASTERISK-28578: race condition on pjsip channelstats command |
ASTERISK-28580: Bypass SYSTEM write permission in manager action allows system commands execution |
ASTERISK-28581: How to integrate gcloud speech recognition with asterisk ivr in python |
ASTERISK-28582: Breaking out of a long playing video when using Background |
ASTERISK-28583: web gui loop |
ASTERISK-28584: Configure direct_media=yes in pjsip. Conf ,don't valid, Media still flows asterisk |
ASTERISK-28585: ari/resource_events: Crash in event session cleanup |
ASTERISK-28586: Typo in README-SERIOUSLY.bestpractices.md |
ASTERISK-28587: Asterisk crash when answering a call |
ASTERISK-28588: MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl |
ASTERISK-28589: chan_sip: Depending on configuration an INVITE can alter Addr of a peer |
ASTERISK-28590: utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument" |
ASTERISK-28591: We have two server with debian os and install asterisk on that, we need server redundancy, how it possible. |
ASTERISK-28592: Asterisk Crash (Segmentation fault) |
ASTERISK-28593: Fix check on forwarded voicemail-to-email message body |
ASTERISK-28594: Chan_SIP Crash (Segmentation fault) |
ASTERISK-28595: Asterisk 15.7.2 with TLS |
ASTERISK-28596: 34/5000 Problem with the sip in asterisk 16 |
ASTERISK-28597: every 2 to 5 mints my asterisk will be stop |
ASTERISK-28598: Configure Fax for receiving in asterisk 13.20 |
ASTERISK-28599: Problem with the sip in asterisk 16 |
ASTERISK-28600: Unable to configure webrtc |
ASTERISK-28601: bridge: BRIDGEPVTCALLID and BRIDGEPEER emtpy after Dial(SIP/...) |
ASTERISK-28602: res_pjsip_outbound_registration: Maximum retries reached |
ASTERISK-28603: Presence subscription on Cisco SIP phone needs special Cisco-styled XML - PJSIP |
ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 |
ASTERISK-28605: chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X |
ASTERISK-28606: Caller Id unknown Showing to receiver |
ASTERISK-28607: P-Asserted-Identity value ${EXTEN} |
ASTERISK-28608: app_amd: Use time calculation to calculate timeout |
ASTERISK-28609: Memory Leak in res_rtp_asterisk.c |
ASTERISK-28610: CDR fields in second leg use wrong variables from first leg |
ASTERISK-28611: sip.conf host!=dynamic peer specific options (e.g. directmedia=off, transport=tcp) not working!? |
ASTERISK-28612: res_pjsip_t38: crash on reinvite with zero port and no c= line |
ASTERISK-28613: func_curl: CURLOPT cannot set Content-Type header |
ASTERISK-28614: app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" |
ASTERISK-28615: chan_dahdi: PRI span status may stay "Down, Active" after a short alarm |
ASTERISK-28616: parking: Deadlock when multi call parking |
ASTERISK-28617: DTMF over SIP INFO in scenarios without audio does not work well |
ASTERISK-28618: bridge_softmix: hold not cleared when joining a softmix bridge |
ASTERISK-28619: data mismatching in "queue show" CLI |
ASTERISK-28620: Segfault in chan_pjsip on pj_strcmp when filtering a transmit message |
ASTERISK-28621: Enforce T.38 error correction mode at 200 ok received |
ASTERISK-28622: Differences in gcc options cause the undefined sanitizer to fail in pjproject when using dev-mode |
ASTERISK-28623: pjsip: PJPROJECT_CONFIGURE_OPTS install location not honored |
ASTERISK-28624: res_pjsip_outbound_registration: add SRV failover |
ASTERISK-28625: Playback of local files impacted by large media cache |
ASTERISK-28626: Missing arguments in PJSIP_CONTACT function documentation |
ASTERISK-28627: Error FRACK!, failed assertion bad magic number |
ASTERISK-28628: Debian 10.2: Warning when app_voicemail is compiling |
ASTERISK-28629: [patch] Add an "inhibitCOLP" flag to the bridges REST API |
ASTERISK-28630: Asterisk crash |
ASTERISK-28631: res_parking: Doesn't park when parkee and parker are the same |
ASTERISK-28632: ConfBridge spawns record_command before MixMonitor ends |
ASTERISK-28633: stasis bridge topic leak |
ASTERISK-28634: Invite loop within PJSIP |
ASTERISK-28635: res_rtp_asterisk should allow for acl style whitelist/blacklist of ICE candidates |
ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR. |
ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. |
ASTERISK-28638: Simplify dialplan for Dial, Page, and ChanIsAvail |
ASTERISK-28639: res_pjsip_endpoint_identifier_ip: Add ability to match on source port |
ASTERISK-28640: app_voicemail with ODBC - error logging is useless |
ASTERISK-28641: res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR |
ASTERISK-28642: SayAlpha not working in GoSub |
ASTERISK-28643: Deadlock, possibly in Parking, maybe in combination with AMI status messages. |
ASTERISK-28644: Stale comment in app_queue about ring_entry exception |
ASTERISK-28645: Menuselect Asterisk |
ASTERISK-28646: asterisk segfault at sp error 4 in libsrtp.so.0.0.0 |
ASTERISK-28647: chan_sip: RTP frames not transmitted after emitting a COLP |
ASTERISK-28648: chan_sip/chan_pjsip copy_via_headers() function not RFC 3261 compliant |
ASTERISK-28649: Segfault: ast_variables_destroy #channel_vars #set_var #sorcery_realtime |
ASTERISK-28650: Voicemail Build Options change not documented |
ASTERISK-28651: chan_sip logs errors on tx to non-existent TCP connections |
ASTERISK-28652: unable to call outbound cli call |
ASTERISK-28653: call through test call file not happening |
ASTERISK-28654: Enabling Real Time Text (RTT) for PJSIP library |
ASTERISK-28655: core: Many things require an audio stream to be present to work |
ASTERISK-28656: improve pjproject.conf sample configuration |
ASTERISK-28657: SIPS TLS connection fails when session ticket extension is used |
ASTERISK-28658: app_confbridge: Add support for setting maximum sample rate |
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them |
ASTERISK-28660: res_fax: wrap Asterisk initiated negotiation with config option |
ASTERISK-28661: chan_iax jitterbuffer growing when time sources not in sync |
ASTERISK-28662: Asterisk install does not compile res_pjsip/config_transport.c |
ASTERISK-28663: jansson: Support old versions |
ASTERISK-28664: "trustrpid" is misspelled in sip_to_pjsip.py |
ASTERISK-28665: No clientcertificate requested |
ASTERISK-28666: Integration with FXO/FXS GAteway via FXO line/OPenvox card |
ASTERISK-28667: Asterisk ignores parsing of config files if a Byte order mark is present |
ASTERISK-28668: aymmetric_rtp_codec dialplan function |
ASTERISK-28669: chan_sip: Device states lost when sip reload |
ASTERISK-28670: astspooldir setting in asterisk.conf isn't configurable |
ASTERISK-28671: Outbound Channel to endpoint that negotiates iLBC mode = 20 produces Warning in CLI and one way audio |
ASTERISK-28672: the sky is falling |
ASTERISK-28673: GET FULL VARIABLE documentation clarification |
ASTERISK-28674: Asterisk becomes unstable after SS7 signalling link restarts |
ASTERISK-28675: Ho we could record both leg of single call separately in wav file format. |
ASTERISK-28676: How we could store DTMF in variable |
ASTERISK-28677: CDR billsec is always 0 for transferred calls |
ASTERISK-28679: stasis application is destroyed after its creation |
ASTERISK-28680: Incorrect results from ast_sorcery_changeset_create() |
ASTERISK-28681: Agent can login queue, but this queue member status is "Invalid" |
ASTERISK-28682: app_record: Lack of `beep` audio file causes application to return error and hangup |
ASTERISK-28683: No clientcertificate requested |
ASTERISK-28684: SIPAddHeader ignored when sending sip MESSAGE |
ASTERISK-28685: check_expr2: linking (when hardening) and cross-compiling troubles |
ASTERISK-28686: chan_sip strictrtp=yes fails when media source is changed: no audio |
ASTERISK-28687: Originate with early media option treats the call as answered which does not reflect the actual state of the call |
ASTERISK-28688: Matching SIP TCP peer by IP with insecure=port regression |
ASTERISK-28689: res_pjsip: Crash when locking group lock when sending stateful response |
ASTERISK-28690: I just want to test something, please ignore |
ASTERISK-28691: unknown codec resulting a call dropped |
ASTERISK-28692: cdr: Asterisk crashed after NoOp application in realtime |
ASTERISK-28693: chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan |
ASTERISK-28694: res_phoneprov: Phone provision stopped updating with a reload of res_phoneprov |
ASTERISK-28695: core: minmemfree watermark uses free RAM, not available RAM |
ASTERISK-28696: PJSIP exception when parsing 'Via' header |
ASTERISK-28697: res_pjsip: Named ACL does not update on reload if changed |
ASTERISK-28698: func_cdr: Getting CDR variables always returns success with a string |
ASTERISK-28699: ast_coredumper does not find asterisk running process and silently fails |
ASTERISK-28700: Page system with music for swimming pool |
ASTERISK-28701: app_queue: Core reload resets queue stats, even when keepstats=yes |
ASTERISK-28702: chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40 |
ASTERISK-28703: PRI-STATUS DOWN |
ASTERISK-28704: AMI QueuePause cannot find interface |
ASTERISK-28705: chan_sip: Phones loose abiltiy to work, core restarting asterisk fixes issue |
ASTERISK-28706: silk 24hHz doesn't show up in 'core show translation' output |
ASTERISK-28707: Pjsip Threadpool cant handle more than 10 calls per second |
ASTERISK-28708: app_queue: Deadlock with "queue show" and "shared_lastcall" option |
ASTERISK-28709: pjproject: Bundled pjproject install error |
ASTERISK-28710: Should be able to disable the /httpstatus URI in the built-in HTTP server |
ASTERISK-28711: HangupCause shows misleading information on timeout |
ASTERISK-28712: Possible freeze in app_queue |
ASTERISK-28713: res_stasis_playback: Error building JSON |
ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults |
ASTERISK-28715: Setting Content-Type header in MessageSend application by using PJSIP |
ASTERISK-28716: ICE: pjnath shouldn't wait for ICE to complete before allowing sending |
ASTERISK-28717: ARI Data Model DialplanCEP - Missing required properties |
ASTERISK-28718: chan_sip: Returns 403 if RTP ports are depleted, should return 503 |
ASTERISK-28719: Cannot remove defaultrule from queue using realtime queues |
ASTERISK-28720: When using realtime queues penaltymemberslimit checks for all members count |
ASTERISK-28721: Asterisk restarts with core dump |
ASTERISK-28722: I agree |
ASTERISK-28723: How do I WebRTC PeerConnection with two Asterisk WebRTC end points |
ASTERISK-28724: AWS Ubuntu : PJSIP call drops after 30 seconds |
ASTERISK-28725: Bridge error on incoming calls on asterisk 16.8.0 and 13.31.0 |
ASTERISK-28726: install_prereq script uses the interactive mode when installing aptitude |
ASTERISK-28727: Some of stasis messages don't contain asterisk_id |
ASTERISK-28728: Asterisk crash in RTP stack (segfault) |
ASTERISK-28729: i can't make a call in my goautodial 4 server |
ASTERISK-28730: res_pjsip_session: Fix out of order session refreshes |
ASTERISK-28731: Directmedia Reinvites have SDP with codecs from configuration not negotiation |
ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls |
ASTERISK-28734: Segmentation fault when calling ast_format_get_codec_id |
ASTERISK-28735: Realtime MoH Unknown format '' -- defaulting to SLIN |
ASTERISK-28736: Asterisk periodic restarts when executing sip reload. |
ASTERISK-28737: Asterisk 13.28.0 repeatable crashes |
ASTERISK-28738: Incorrect state machine used when MOH_PASSTHRU is used |
ASTERISK-28739: Dropping redundant connected line update |
ASTERISK-28740: Issue regarding Sip account registeration |
ASTERISK-28741: Issue regarding Sip account registeration |
ASTERISK-28742: res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup |
ASTERISK-28743: Asterisk is crashing if the 200 OK with SDP |
ASTERISK-28744: No transfer events logged in queue_log |
ASTERISK-28745: [BOUNTY] support_path missing after reload |
ASTERISK-28746: res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set |
ASTERISK-28747: YES/NO attributes are not set properly when creating PJSIP sorcery objects via ARI |
ASTERISK-28748: Recording failed when making many calls per second |
ASTERISK-28749: Matching on Caller ID not working if the dialled extension is a pattern |
ASTERISK-28750: TLS/SSL Key too small error |
ASTERISK-28751: Difficulty sending 16k slin16 back to Asterisk via External Media |
ASTERISK-28752: Support receiving both t38 and t30 faxes |
ASTERISK-28754: ASTERISK-28738 Causes Audio Issue After Hold |
ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables" field |
ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option |
ASTERISK-28757: 403 - Forbidden auth ID when SIP messaging Anveo |
ASTERISK-28758: pjsip startup errors when using "with-ssl" configure option |
ASTERISK-28759: A non negotiated rtp frame causes call disconnection when there is a SSRC change |
ASTERISK-28760: G729a codec can't be loaded |
ASTERISK-28761: Assigning CallerIDNum from DNIDDigits to chan_pjsip |
ASTERISK-28762: Problem setting up ssl connection. Internal SSL error |
ASTERISK-28763: Issue with SQLBindParameter with ODBC on StrLen_or_IndPtr |
ASTERISK-28764: res_rtp_asterisk: Improve NACK support and seqno handling |
ASTERISK-28765: tcptls API: bad file descriptor when connection fails |
ASTERISK-28766: PJSIP blind transfer not completed after using Proceeding() |
ASTERISK-28767: chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late |
ASTERISK-28768: No audio on remote NATted phone when using local_net behind another NATted Asterisk |
ASTERISK-28769: DTLS Handshake Fails to Occur if ice_support is enabled but not used |
ASTERISK-28770: res_pjsip: AVC denial with default SELinux setup on CentOS 7 |
ASTERISK-28771: Unable to install asterisk in Ubuntu 12/14/16/18.04 versions |
ASTERISK-28772: Add indication tones for Indonesia |
ASTERISK-28773: Incorrect Sender SSRC in RTCP when p2p rtp bridge is active |
ASTERISK-28774: chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge |
ASTERISK-28775: SRV fix for ASTERISK-28746 fails with Deutsche Telecom |
ASTERISK-28776: Non async-signal-safe syscalls used after fork before exec |
ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option |
ASTERISK-28778: Public IP in contact URI when softphone traffic goes through VPN |
ASTERISK-28779: what will be the process to send dtmf after call bridge using agi. |
ASTERISK-28780: app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup |
ASTERISK-28781: res_config_odbc: Sorcery doesn't set default value if the ODBC realtime value is blank |
ASTERISK-28782: Add support for Content-Disposition header in multi-part INVITES |
ASTERISK-28783: res_pjsip_session: Allow default non-audio streams to have reflected state |
ASTERISK-28784: res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream |
ASTERISK-28785: chan_local: Unnecessary transcoding for Originate (local channels) |
ASTERISK-28786: Duplicated Endpoints, AORS, Auths user ARI sorcery |
ASTERISK-28787: res_pjsip_session: Decide more intelligently when to add video |
ASTERISK-28788: func_aes: incorrectly printing error 'declined to load' |
ASTERISK-28789: test_utils: incorrectly printing error 'declined to load' |
ASTERISK-28790: Crash during conference call using confbridge and video |
ASTERISK-28791: Manager Action MixMonitorMute not working |
ASTERISK-28792: codec_gsm: while building, optimization flag is overwritten |
ASTERISK-28793: Asterisk 13.32.0 crash in pjsip_tx_data_add_ref |
ASTERISK-28794: res_pjsip: Crash when escaping during URI printing |
ASTERISK-28795: channel: write to a stream on multi-frame writes |
ASTERISK-28796: func_channel: cannot read fields exten, context, userfield, channame from dialplan |
ASTERISK-28797: [patch] tcptls: Fix notice when TLS is enabled but not configured. |
ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server. |
ASTERISK-28799: I have issue |
ASTERISK-28800: core: SIGSEGV on DTMF when some modules not loaded |
ASTERISK-28801: [patch] stasis: Avoid always true warnings with clang. |
ASTERISK-28802: ari: /dial function disconnect Inbound call automatically if Outbound is "busy" or "discard" in case delay "Answer" message. |
ASTERISK-28803: [patch] chan_unistim: Avoid tautological warnings with clang. |
ASTERISK-28804: [patch] app_osplookup.c: Avoid a format truncation. |
ASTERISK-28805: follow me with empty database fileds broken after res odbc fix for empty strings |
ASTERISK-28807: ICES dialplan issue |
ASTERISK-28808: [patch] test_stasis: Avoid always true warning with clang. |
ASTERISK-28809: [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction. |
ASTERISK-28810: Segmentation fault in ast_manager_build_channel_state_string_prefix |
ASTERISK-28811: Crash occurs when fax session switches from T.38 to audio |
ASTERISK-28812: First DTMF is not get |
ASTERISK-28813: func_volume: Allow decimal numbers as parameter to improve granularity |
ASTERISK-28814: Asterisk stops processing SIP requests because of an undetermined reason |
ASTERISK-28815: res_pjsip.so segfaulting on 17.3.0 |
ASTERISK-28816: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers. |
ASTERISK-28817: chan_pjsip: constant DTMF tone if RTP is not setup yet |
ASTERISK-28818: [patch] BuildSystem: Allow space in path. |
ASTERISK-28819: [patch] bridge_softmix_binaural: Show state in menuselect. |
ASTERISK-28820: pjproject_bundled: Modes Developer+Noisy give Stop. |
ASTERISK-28821: a code change to chan_pjsip breaks SIP/ISUP internetworking in early state |
ASTERISK-28822: chan_unistim.c: Recv error 9 (Bad file descriptor) |
ASTERISK-28823: Updates for outgoing registrations not sent to the correct network address |
ASTERISK-28824: BuildSystem: Search for Python/C API when possibly needed only. |
ASTERISK-28825: Any curl response checks out as valid even if 404 is returned. |
ASTERISK-28826: res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK |
ASTERISK-28827: res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK |
ASTERISK-28828: NOTIFY sequence upsets MS Teams SIP trunk |
ASTERISK-28829: app_queue: leaking stasis subscription when Redirecting call |
ASTERISK-28830: Incorrect UTF-8 handling when using function FILTER |
ASTERISK-28831: Leaking stasis subscriptions can linger indefinitely and brick Asterisk |
ASTERISK-28832: chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio |
ASTERISK-28833: SIP Hang- All the SIP peers are going unreachable at same time. |
ASTERISK-28834: Segfault in taskprocessor_push |
ASTERISK-28835: IPv6 addresses in SDP incorrectly formatted |
ASTERISK-28836: chan_oss: Video Console broken. |
ASTERISK-28837: pjproject_bundled: Honor --without-pjproject. |
ASTERISK-28838: AST_MODULE_INFO requires, MODULEINFO does not mention |
ASTERISK-28839: Sporadic crashes with Segmentation fault |
ASTERISK-28840: samples: Missing for the modules cdr_ and cel_sqlite3_custom. |
ASTERISK-28841: app_confbridge: Add support for disabling text messaging for a user |
ASTERISK-28842: Hello |
ASTERISK-28843: res_rtp_asterisk: Duplicate detection of DTMF - Wideband codec / MS customisation |
ASTERISK-28844: samples: Start and Reload are blathering. |
ASTERISK-28845: segfault and then crash |
ASTERISK-28846: stream: Enforce formats immutability |
ASTERISK-28847: ARI channels cuts the endpoint string over 80 characters |
ASTERISK-28848: app_fax: Compile. |
ASTERISK-28849: asttest failed to compile on fedora |
ASTERISK-28850: sipp: Non-compliant XML files do not work with 3.6.0 |
ASTERISK-28851: Add useragent to CLI > 'pjsip show contacts' |
ASTERISK-28852: Unprotected access to nochecksums variable, causes build failures |
ASTERISK-28853: Missing include on FreeBSD |
ASTERISK-28854: SIGSEGV when pjsip show history encounters IPV6 address |
ASTERISK-28855: PJSIP - Implement CHANNEL(secure_bridge_media) |
ASTERISK-28856: Codec Negotiation: Add incoming_call_answer_pref and outgoing_call_answer_pref |
ASTERISK-28857: #exec call in pjsip.conf causes large delay (60 seconds) when reloading pjsip, partially locks up dialplan. |
ASTERISK-28858: app_queue: Realtime linear queues losing the order of agents |
ASTERISK-28859: pjsip: Increase maximum candidate count |
ASTERISK-28860: pjsip: Resolve unsolicited->solicited aggregate issue |
ASTERISK-28861: testsuite: Resolve unsolicited->solicited non-aggregate issue |
ASTERISK-28862: res_musiconhold: Race condition between starting/stopping |
ASTERISK-28863: The ast_rtp_codecs_payloads functions don't preserve order |
ASTERISK-28864: RTP Timestamp not increasing after several transfers and codec changes |
ASTERISK-28865: FAX T.38 re-Invite failed on '491 Another INVITE transaction in progress' |
ASTERISK-28866: third-party/pjproject/configure.m4 contains bashisms |
ASTERISK-28867: cannot get ANSWER Status from ${DIALSTATUS} though i get busy congested |
ASTERISK-28868: app_alarmreceiver: does not call "eventcmd" with events as arguments or piping |
ASTERISK-28869: pjsip: Crash in timer when sending request |
ASTERISK-28870: streams: One memory leak and one issue cloning streams |
ASTERISK-28871: res_pjsip_session: Unnecessary re-Invite on call answer |
ASTERISK-28872: Asterisk services are not starting with ssl on 8089 port after configured http.conf |
ASTERISK-28874: res_rtp_asterisk: RFC2833/RFC4733 Minimum signal duration not adhered to |
ASTERISK-28875: All sip tranks are regularly unregister |
ASTERISK-28876: Wrong next hop for INVITEs with PJSIP and PATH |
ASTERISK-28877: Duplicate RINGING DeviceStateChange AMI events from chan_sip.c |
ASTERISK-28878: chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 |
ASTERISK-28879: pjproject has race conditions in it's build system |
ASTERISK-28880: res_xmpp: Does not support urn:xmpp:ping causing session termination |
ASTERISK-28881: res_pjsip_pubsub: Option to reduce log verbosity by selectively disabling missing/invalid subscriptions |
ASTERISK-28882: res_pjsip: Contact not completely removed on transport closure |
ASTERISK-28883: Spyee information ist missing in ChanSpyStop AMI Event |
ASTERISK-28884: x-ast-orig-host not filtered out from request URI and To header |
ASTERISK-28885: res_rtp_asterisk: Simultaneous termination and ICE complete can cause crash |
ASTERISK-28886: chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2 |
ASTERISK-28887: sendFAX fail with T.38 |
ASTERISK-28888: res_corosync: causes asterisk crash in huge distributed environment. |
ASTERISK-28889: Outbound calls drops |
ASTERISK-28890: res_pjsip_sdp_rtp: Keepalive not supported for video streams |
ASTERISK-28891: documentation: AGICommand_set+music documentation arguments displayed incorreclty |
ASTERISK-28892: res_musiconhold: Module res_musiconhold throws false warning |
ASTERISK-28893: pbx_realtime: Cascading deadlock due to ast_autoservice_stop blocking |
ASTERISK-28894: new memory leak when updating 16.9.0 to 16.10.0 |
ASTERISK-28895: res_pjsip_logger: Add tons'o'functionality |
ASTERISK-28896: ari: Add support for specifying variables on channel create |
ASTERISK-28897: app_confbridge: AMI Event "ConfbridgeTalking off" not fired when user leaves ConfBridge while talking |
ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp packets |
ASTERISK-28899: Upgrade Asterisk to bundled pjproject 2.10 |
ASTERISK-28900: res_fax: Double frame free when gateway in use with off-nominal format usage |
ASTERISK-28901: pjsip behaves incorrectly when sending RTP, it sends it to a private IP |
ASTERISK-28902: High Memory consumption |
ASTERISK-28903: res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. |
ASTERISK-28904: RTP ICE leaks the memory |
ASTERISK-28905: Wrong Download Link On https://www.asterisk.org/download-asterisk-thank-you |
ASTERISK-28906: ari: Race condition when destroying holding bridge multiple times with channels in it |
ASTERISK-28907: Crash when removing stasis topic |
ASTERISK-28908: pjsip_message_filter: 400 'Missing Contact header' reply to wrong port |
ASTERISK-28909: chan_pjsip: CLI 'pjsip show channelstats' shows fax T.38 session as 'not valid' and doesn't print video stream stats |
ASTERISK-28910: PJSIP: invalid value error exception when parsing 'Contact' header |
ASTERISK-28911: Segmentation Fault on Voicemail Menu |
ASTERISK-28912: Bug with a 'core show channels' |
ASTERISK-28913: How to have digits spelled in different language |
ASTERISK-28914: res_http_websocket: Client doesn't use mask |
ASTERISK-28915: phone number porting |
ASTERISK-28916: Memory leak with Asterisk 16 and malformed REGISTER requests |
ASTERISK-28917: sip peers become unreachable |
ASTERISK-28918: No Application SIPAddHeader() |
ASTERISK-28919: we have been able to setup asterisk webrtc video call on both the endpoints but there is no audio from caller end but audio at receiver's end is working fine |
ASTERISK-28920: bridge show all causes crash |
ASTERISK-28921: Wrong return value check for fwrite when writing to pcap file |
ASTERISK-28922: Attended transfer not swapping channel |
ASTERISK-28923: T.38 Segfaults in chan_pjsip_queryoption |
ASTERISK-28924: Imposible to add or read sip headers from a 302 Redirect |
ASTERISK-28925: memory leak in asterisk 16.10 |
ASTERISK-28926: core dump trying to free null channel snapshot |
ASTERISK-28927: Asterisk crash in music on hold |
ASTERISK-28928: message proxy changes when client login with different IP |
ASTERISK-28929: pjproject_bundled: Honor --without-pjproject. |
ASTERISK-28930: ./configure --without-ssl build failure |
ASTERISK-28931: stasis.c:1475 publish_msg: FRACK!, Failed assertion bad magic number 0x0 for object |
ASTERISK-28932: res_pjsip_logger writing too big packets |
ASTERISK-28933: res_pjsip.so fails to load when bundled pjproject is compiled without libssl |
ASTERISK-28934: chan_mobile won't load without chan_alsa |
ASTERISK-28935: app_meetme tries to create audio files with format sln |
ASTERISK-28936: res_pjsip: crash when dialing non-sip uri |
ASTERISK-28937: Task processor queue reached 500 scheduled tasks |
ASTERISK-28938: core_unreal / core_local: Add support for multistream and re-negotiation |
ASTERISK-28939: res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC |
ASTERISK-28940: /channels/create doesn't get any parameters from the body |
ASTERISK-28941: segfault in pjsip timer |
ASTERISK-28942: res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching |
ASTERISK-28943: Asterisk can't start with the errors media_cache_item_populate_from_astdb: Unable to obtain information for file /tmp/... |
ASTERISK-28944: bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation |
ASTERISK-28945: AMI SendText - add Content-Type parameter |
ASTERISK-28946: Missing/wrong endpoint default values in configInfo documentation |
ASTERISK-28947: Segmentation fault in mixmonitor_ds_destroy |
ASTERISK-28948: ARI channel create doesn't referencing the channel_id parameter |
ASTERISK-28949: res_http_websocket: Add masking to websocket client |
ASTERISK-28950: Stale code in app_queue to check untouched channel |
ASTERISK-28951: Inconsistent behaviour queues.conf when there is (not) a [general] section |
ASTERISK-28952: Queue wrapuptime sometimes not respected (based on stale lastcall time) |
ASTERISK-28953: res_pjsip_session: Preserve stream label |
ASTERISK-28954: StreamEcho() only returns 1 active stream |
ASTERISK-28955: "setvar" doesn't work properly in dahdi-channels.conf |
ASTERISK-28956: res_odbc: ODBC connection does not always reconnect |
ASTERISK-28957: chan_sip: chan_sip does not process 400 response to an INVITE. |
ASTERISK-28958: Continue reading string when ping received by websocket |
ASTERISK-28959: res_pjsip: Added option for disable rport parameter set |
ASTERISK-28960: bridge: System gets into state where bridge is terminated after joining |
ASTERISK-28961: res_pjsip_outbound_registration: Re-registration is incorrectly timed when response contains two identical Contact headers |
ASTERISK-28962: Asterisk Memory Leak after SIP Reply Flood |
ASTERISK-28963: Asterisk is killed when I connected to AMI |
ASTERISK-28964: How to increase the ringback time of SIP endpoints registered to Asterisk |
ASTERISK-28965: res_pjsip: Apply outbound proxy to static contacts on AOR |
ASTERISK-28966: chan_pjsip: Interaction with provider re-INVITE and RTP causes codec flip-flop |
ASTERISK-28967: Increase Agent Timeout in Queue |
ASTERISK-28968: res_pjsip: Crash when comparing header in outgoing SIP request |
ASTERISK-28969: res_pjsip: AMI command for show endpoint does not reflect active channels |
ASTERISK-28970: Reflected XSS |
ASTERISK-28971: app_userevent: Does not handle non-ASCII characters |
ASTERISK-28972: FRACK! + task processor queue issue |
ASTERISK-28973: Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) |
ASTERISK-28974: res_rtp_asterisk: T.140 messages have appended RTP string to each message block. |
ASTERISK-28975: res_http_websocket: Text payload data doesn't necessary include trailing zero |
ASTERISK-28976: Crashes releted to pjsip |
ASTERISK-28977: PJSIP can't SET CallerId |
ASTERISK-28978: acl: named_acl rule misconfiguration results in segfault on reading rule from realtime |
ASTERISK-28979: Not able to make call using PHP Agi |
ASTERISK-28980: PJSIP outbound registration issue |
ASTERISK-28981: I meet errors as I follow the pjsip setup scenario, how to fix it? |
ASTERISK-28982: res_pjsip_t38: Does not resume as audio when negotiation fails |
ASTERISK-28983: Unable to redirect outgoing calls to mobile |
ASTERISK-28984: Asterisk suddenly crashs |
ASTERISK-28985: ari: Sends channel event to multiple applications when connecting multiple over same socket |
ASTERISK-28986: video over audio is not switching in webrtc with asterisk 16 |
ASTERISK-28987: BridgeCreated ARI event shows wrong video_mode info |
ASTERISK-28988: Asterisk is crashing when we receive incoming calls |
ASTERISK-28989: Asterisk is crashing when we receive incoming calls |
ASTERISK-28990: chan_pjsip: Device state does not reflect hold |
ASTERISK-28991: bridging: No channel is present when writing action |
ASTERISK-28992: app_voicemail: Deadlock in ODBC when retrieving file |
ASTERISK-28993: res_pjsip: Wrong Via and Contact is chosen, despite explicit configured transport |
ASTERISK-28994: Limit Prefix |
ASTERISK-28995: res_pjsip_registrar: Expires on statically configured contacts is not correct |
ASTERISK-28996: chan_sip: TLS - Bad file descriptor errors |
ASTERISK-28997: res_pjsip: Asterisk locks up and stops processing any SIP requests |
ASTERISK-28998: Segfault in pj/timer.c |
ASTERISK-28999: pjsip / sorcery / ao2: Large refcounts (FRACKs) on reload of large pjsip.conf configuration |